In this description, the term sending communication device refers to a communication device including a transmitter being arranged to send multimedia streams to a communication network. The term receiving communication device refers to a communication device including a receiver for receiving multimedia streams from the communication network, respectively. It is obvious that the same communication device may include both the transmitter and the receiver whereby allowing one-way or two-way communication with the communication network. A wireless communication device includes a transmitter and/or a receiver implementing wireless communication in a wireless communication network. The term wireless communication system, such as a mobile communication system, generally refers to any communication system which makes a wireless data transmission connection possible between a wireless communication device and stationary parts of the system, the user of the wireless communication device moving within the operating range of the system. A typical wireless communication system is a Public Land Mobile Network (PLMN). A well-known example is the GSM system (Global System for Mobile telecommunications). The invention preferably relates to the third generation of mobile communication systems. As an example, the Universal Mobile Telecommunications System (UMTS) is used as an example of such a third-generation communication system.
In third generation systems, the terms bearer service and service are used. A bearer service is a telecommunication service type which provides the facility to transmit signals between access points. In general, the bearer service corresponds to the term of a traffic channel which defines, for example, the data transmission rate and the quality of service (QoS) to be used in the system when information is transmitted between a wireless communication device and another part of the system. The bearer service between the wireless communication device and the base station is, for example, a radio bearer service, and the bearer service between the radio network control unit and the core network is, for example, an lu bearer service (Interface UMTS bearer). In the UMTS system, the interface between the radio network control unit and the core network is called lu interface. In UMTS there is also the so called GERAN part, which uses, in addition to the lu interface, also an interface called as Gb interface. In this connection, the service is provided by the mobile communication network for performing a task (tasks); for example, data services perform data transmission in the communication system, telephone services are related to telephone calls, multimedia, etc. Thus, the service requires data transmission, such as a telephone call or the transmission of multimedia streams, between the wireless communication device and the stationary parts of the system. One important task of the operation of a third-generation mobile communication system is to control (initialize, maintain and terminate, according to the need) bearer services in such a way that each requested service can be allocated to mobile stations without wasting the available bandwidth.
The quality of service determines, for example, how protocol data units (PDU) are processed in the mobile communication network during the transmission. For example, QoS levels defined for connection addresses are used for controlling the transmission order, buffering (packet strings) and rejecting packets in support nodes and gateway support nodes, particularly when two or more connections have packets to be transmitted simultaneously. The different QoS levels determine, for example, different delays for packet transmissions between the different ends of the connection, as well as different bit rates. Also, the number of rejected and/or lost packet data units may vary in connections with different QoS levels.
It is possible to request for a different QoS for each PDP context. For example, in e-mail connections, a relatively long delay can be allowed in the transmission of streams. However, real-time interactive applications, such as video conferencing, require packet transmission at a high rate. In some applications, such as file transfers, it is important that the packet switched transmission is faultless, wherein in error situations, the packet data units are retransmitted, if necessary.
For the packet switched communication service in the UMTS system, the defining of four different traffic classes has been proposed, and for the properties of these traffic classes, the aim has been to consider the different criteria for the different connection types. One criterion defined for the first and second classes is that the transmission takes place in real time, wherein the transmission must have no significant delays. However, in such classes, the accuracy of the data transfer is not such an important property. In a corresponding manner, non-real time data transmission is sufficient for the third and fourth traffic classes, but a relatively accurate data transmission is required of them. An example of real-time first-class communication is the transmission of conversational speech signals in a situation in which two or more persons are discussing with each other by means of wireless communication devices. An example of a situation in which real-time second-class communication might be feasible, is the transmission of a video signal for immediate viewing (streaming). Third-class non-real time packet communication can be used, for example, for the use of database services, such as the browsing of Internet home pages, in which the relatively accurate data transmission at a reasonable rate is a more important factor than the real-time data transmission. In the system according to this example, for example the transfer of e-mail messages and files can be classified to the fourth category. Naturally, the number of traffic classes is not necessarily four as mentioned here, but the invention can be applied in packet switched communication systems comprising any number of traffic classes. The properties of the four presented traffic classes are briefly presented in Table 1.
TABLE 1ClassThird classFourth class(interactive(backgroundclass):class):First classSecond classinteractivebackground(conversational(streaming class):best efforttransmission byclass):real-time, e.g.methodthe best effortreal-time, e.g.video informationacknowledgementmethodtelephoneguaranteedInternetacknowledgementconversationcapacitybrowser,backgroundguaranteedacknowledgementTelnetloading of e-mailcapacitypossiblereal-timemessages,nobuffering oncontrolcalendar events,acknowledgementapplication levelchanneletc.Maximum<2048<2048<2048-overhead<2048-overheadbit rate(kbps)DeliveryYes/NoYes/NoYes/NoYes/NoorderMaximum≦1500 or 1502≦1500 or 1502≦1500 or≦1500 or 1502packet1502size(Bytes)(SDU)TransmissionYes/No/—Yes/No/—Yes/No/—Yes/No/—ofincorrectpackets(SDU)Residual5 * 10−2, 10−2, 5 * 10−3,5 * 10−2, 10−2,4 * 10−3, 10−5,4 * 10−3, 10−5,bit error10−3, 10−4, 10−5,5 * 10−3, 10−3, 10−4,6 * 10−86 * 10−8ratio10−610−5, 10−6Packet10−2, 7 * 10−3, 10−3,10−1, 10−2, 7 * 10−3,10−3, 10−4, 10−610−3, 10−4, 10−6error ratio10−4, 10−510−3, 10−4,(SDU)10−5Transmission100 ms-maximum value250 ms-maximum valuedelay (ms)Guaranteed<2048<2048bit rate(kbps)Traffic1, 2, 3processingpriorityAllocation1, 2, 31, 2, 31, 2, 31, 2, 3priority
The guaranteed bit rate is used for admission control and resource reservation at the RAN and CN, the maximum bit rate is used for policing at the CN, i.e. no higher than the maximum bit rate is allowed to enter the CN at the GGSN, packets that exceed this bit rate will be dropped.
Modern second and third generation wireless communication devices have much better data processing properties than older wireless communication devices. For example, they already have the facility of connecting to the Internet and using a browsing application in the wireless communication device to retrieve information from the Internet, and in the future, it will be possible to set up multimedia calls, for example, for real-time video conferences and the like.
The requirements of different applications may be significantly different. Some applications require fast communication between the sender and the receiver. These applications include, for example, video and telephone applications. Some other applications may require as accurate data transmission as possible, but the bit rate of the data transmission connection is less important. These applications include, for example, e-mail and database applications. On the other hand, these applications can be used in several wireless communication devices with different properties.
The user of the wireless communication device may be willing to watch a multimedia presentation with the wireless communication device. The user finds the loading address of such a presentation and sends a request to send the presentation to the wireless communication device. The request is handled in the communication system. The loading address of the requested multimedia presentation may address to a server in a communication network, such as a server of the Internet. The server which delivers the multimedia presentation to the receiving wireless communication device is called as a streaming server in this description.
The communication system should reserve enough resources for the communication between the streaming server and the wireless communication device to be able to deliver the requested multimedia presentation. Otherwise the presentation may not be presented with the same accuracy and error free in the receiving wireless communication device. In the UMTS communication system the wireless communication device requests a PDP context with certain QoS parameters first. Then, the network selects a bearer for the connection by using some selection bases, for example, the parameters the wireless communication device has possibly used in the request. Such selection bases may not be appropriate or accurate enough wherein situations may occur in which the bearer service can not provide enough transmission capacity for the connection, or it provides more capacity than is needed, wherein the usage of the network resources is not efficient.
Another situation in which a delivery of multimedia information may be needed is two wireless communication devices communicating with each other to exchange multimedia information such as video or still images. Also in this kind of situation enough resources should be reserved by the network for the communication. However, when using prior art methods it is not always possible to inform both ends of the connection about the demands for the connection.
Basic streaming systems are non-adaptive. For example, the current Packet Switched streaming Service (PSS) defined by 3GPP in releases 4 and 5 is non-adaptive. Packet Switched streaming Service in Rel. 6 will be adaptive. The adaptive characteristic is given by the ability of the system, i.e. both a streaming server and a client, to adapt to the varying network channel conditions such as changes in the QoS negotiated channel bit rates, transfer delays, other Quality of Service parameters, or even changes in the underlying network in case of handovers.
In order to make the system adaptive, some communication between the streaming server and clients must be established. This is already in place whenever the RTSP protocol is used for session set-up and control. However, the transmission of the necessary information between the server and the client must occur in a correct way in order to guarantee that the system is adaptive and ultimately the best user Quality of Service for audio and video streaming can be achieved.
For this purpose, some prior art techniques already enable the transmission of QoS information, coming from the underlying mobile network, from a streaming client to a streaming server. This allows more cooperation between the two ends in order to make the system more adaptive.
What has not been specified so far is the relation between the QoS parameters in a specific mobile network environment and the PDP (Packet Data Protocol) context usage. For instance, different cases are possible. In the following, the associated RTCP flow related to each RTP media stream is not considered. Alternatively, considering the RTP and its associated RTCP flow as a part of the same multimedia stream does not change the nature of the problem:    1. A PDP context carries only one media of a streaming session    2. A PDP context carries all the media of a streaming session in a case when there is more than one media.
If the streaming client decides to signal to the streaming server e.g. via RTSP some of the QoS profile parameters, for example the guaranteed bit rate, the maximum bit rate or the transfer delay, some problems may occur to the server in the correct interpretation of the QoS profile and, in the end, in the nature of the network connection.
In RTSP there are two possible kinds of sessions, which are a so called aggregate controlled session and a non-aggregate controlled session. The aggregate controlled session is a session where, at the transport level, all media components can be controlled by a single command sent to the server by the client (e.g. one RTSP PLAY command for both audio and video components). If this does not happen, i.e. at least one media component is controlled individually in a session, then the session is said to have non-aggregate control.
In the following, some examples are disclosed to clarify the problems which relate to the negotiation of QoS parameters for multimedia streams. It should be noted that the examples and the different parameters used in the examples are non-restrictive and in practical implementations different kind of parameters and combinations of media streams may exist.